• The KIPS Transactions:PartC
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• v.12C no.1 s.97
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• pp.121-128
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• 2005

Tho existing TCP(Transmission Control Protocol) is known to be unsuitable for a network with the characteristics of high RDP(Bandwidth-Delay Product) because of the fixed small or large buffer size at the TCP sender and receiver. Thus, some trial cases of adjusting the buffer sizes automatically with respect to network condition have been proposed to improve the end-to-end TCP throughput. ATBT(Automatic TCP fluffer Tuning) attempts to assure the buffer size of TCP sender according to its current congestion window size but the ATBT assumes that the buffer size of TCP receiver is maximum value that operating system defines. In DRS(Dynamic Right Sizing), by estimating the TCP arrival data of two times the amount TCP data received previously, the TCP receiver simply reserves the buffer size for the next arrival, accordingly. However, we do not need to reserve exactly two times of buffer size because of the possibility of TCP segment loss. We propose an efficient TCP buffer tuning technique(called TBT-PLR: TCP buffer tuning algorithm based on packet loss ratio) since we adopt the ATBT mechanism and the TBT-PLR mechanism for the TCP sender and the TCP receiver, respectively. For the purpose of testing the actual TCP performance, we implemented our TBT-PLR by modifying the linux kernel version 2.4.18 and evaluated the TCP performance by comparing TBT-PLR with the TCP schemes of the fixed buffer size. As a result, more balanced usage among TCP connections was obtained.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.17 no.11
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• pp.2553-2562
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• 2013

It is difficult for TCP congestion control algorithm to ensure the bandwidth and delay bound required for media streaming services in broadband wireless network environments. In this paper, we propose the COIN TCP (COncave INcrease TCP) scheme for providing a high-quality media streaming services. The COIN TCP concavely increases the congestion window size by adjusting the increment rate of congestion window, that is inversely proportional to the amount of data accumulated in the router queue. As a result, our scheme can quickly occupy the available bandwidth and prevent the heavy congestion. It also improves the link utilization by adjusting the decrement rate of congestion window according to the packet loss rate with the random loss. Through the simulation results, we prove that our scheme improves the total throughput in broadband wireless network.

• Journal of Advanced Navigation Technology
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• v.11 no.1
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• pp.31-36
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• 2007

TCP (Transmission Control Protocol) is one of the protocols which are widely used from the Internet environments. Through the flow control of TCP, it could be increased efficiency for the loss and a re-transmission of data and the flow control become accomplished through window technique which puts the limit of size. By the flow control, TCP divided in various versions. In this paper, it is analyzed the simulation result which applies the error model in the Newreno which is an improved model of the representative Tahoe, Reno.

• International Journal of Internet, Broadcasting and Communication
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• v.16 no.1
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• pp.29-35
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• 2024

TCP is considered as one of the major candidate transport protocols even for constrained IoT networks..In our previous work, we investigated the congestion control mechanism of the uIP TCP. Since the uIP TCP sets the window size to one segment by default, managing the retransmission timer is the primary approach to congestion control. However, the original uIP TCP sets the retransmission timer based on the fixed RTO, it performs poorly when a radio duty cycling mechanism is enabled and the hidden terminal problem is severe. In our previous work, we proposed a TCP retransmission timer adjustment scheme for uIP TCP which adopts the notion of weak RTT estimation of CoCoA, exponential backoffs with variable limits, and dithering. Although our previous work showed that the proposed retransmission timer adjustment scheme can improve performance, we observe that the scheme often causes a node to set the retransmission timer for an excessively too long time period. In this work, we show that slightly modifying the dithering mechanism of the previous scheme is effective for improving TCP fairness.

• Journal of Information Processing Systems
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• v.2 no.2
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• pp.76-81
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• 2006

A PDA is used mainly for downloading data from a stationary server such as a desktop PC in an infrastructure network based on wireless LAN. Thus, the overall performance depends heavily on the performance of such downloading with PDA. Unfortunately, for a PDA the time taken to receive data from a PC is longer than the time taken to send it by 53%. Thus, we measured and analyzed all possible factors that could cause the receiving time of a PDA to be delayed with a test bed system. There are crucial factors: the TCP window size, file access time of a PDA, and the inter-packet delay that affects the receiving time of a PDA. The window size of a PDA during the downstream is reduced dramatically to 686 bytes from 32,581 bytes. In addition, because flash memory is embedded into a PDA, writing data into the flash memory takes twice as long as reading the data from it. To alleviate these, we propose three distinct remedies: First, in order to keep the window size at a sender constant, both the size of a socket send buffer for a desktop PC and the size of a socket receive buffer for a PDA should be increased. Second, to shorten its internal file access time, the size of an application buffer implemented in an application should be doubled. Finally, the inter-packet delay of a PDA and a desktop PC at the application layer should be adjusted asymmetrically to lower the traffic bottleneck between these heterogeneous terminals.

• Journal of Communications and Networks
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• v.11 no.5
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• pp.518-528
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• 2009

IEEE 802.11 wireless local area network (WLAN) has become the prevailing solution for wireless Internet access while transport control protocol (TCP) is the dominant transport-layer protocol in the Internet. It is known that, in an infrastructure-based WLAN with multiple stations carrying long-lived TCP flows, the number of TCP stations that are actively contending to access the wireless channel remains very small. Hence, the aggregate TCP throughput is basically independent of the total number of TCP stations. This phenomenon is due to the closed-loop nature of TCP flow control and the bottleneck downlink (i.e., access point-to-station) transmissions in infrastructure-based WLANs. In this paper, we develop a comprehensive analytical model to study TCP dynamics in infrastructure-based 802.11 WLANs. We calculate the average number of active TCP stations and the aggregate TCP throughput using our model for given total number of TCP stations and the maximum TCP receive window size. We find out that the default minimum contention window sizes specified in the standards (i.e., 31 and 15 for 802.11b and 802.11a, respectively) are not optimal in terms of TCP throughput maximization. Via ns-2 simulation, we verify the correctness of our analytical model and study the effects of some of the simplifying assumptions employed in the model. Simulation results show that our model is reasonably accurate, particularly when the wireline delay is small and/or the packet loss rate is low.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.6B
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• pp.430-439
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• 2008

In this paper, we propose a sender-based TCP congestion control scheme for downward vertical handover (DVHO), in which mobile node moves from a cellular network to a wireless LAN. DVHO can give rise to severe performance problems in TCP throughput because it causes a drastic change of link characteristics. Particularly, TCP executes falsely congestion control by packet reordering, which is occurred from link delay difference between a cellular link and a wireless LAN link. Therefore, the congestion window is reduced. And unnecessary retransmissions wastes bandwidth. To solve these problems, we propose a method using estimated round-trip time in cellular link to process duplicated ACKs from reordering. Furthermore, the duplicated ACKs are used to the control congestion window size. Simulation result shows that the proposed scheme can solve problems. Moreover, the proposed scheme can have better performance than TCP New Reno and nodupack.

• Journal of Satellite, Information and Communications
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• v.8 no.1
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• pp.1-7
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• 2013

Currently satellite communication systems usually use the ACM(Adaptive Coding and Modulation) to extend the link availability and to increase the bandwidth efficiency. However, when ACM system is used for satellite communications, we should carefully consider TCP congestion control to avoid network congestions. Because MODCODs in ACM are changed to make a packet more robust according to satellite wireless link conditions, bandwidth of satellite forward link is also changed. Whereas TCP has a severe problem to control the congestion window for the changed bandwidth, then packet overflow can be experienced at MAC or PHY interface buffers. This is a reason that TCP in transport layer does not recognize a change of bandwidth capability form MAC or PHY layer. To overcome this problem, we propose the adaptive congestion control scheme of TCP for supporting ACM in Satellite PEP (Performance Enhancing Proxy) systems. Simulation results by using ns-2 show that our proposed scheme can be efficiently adapted to the changed bandwidth and TCP congestion window size, and can be useful to improve TCP performance.

• The KIPS Transactions:PartC
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• v.8C no.3
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• pp.351-358
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• 2001

무선망에서는 유선망에 비해 그 특성상 비교적 많은 패킷을 손실된다. TCP 프로토콜은 흐름제어나 에러정정, 혼잡 제어 등의 기능을 통해 보다 효율적이고 안정적인 통신을 지원하고 있다. 하지만 표준 TCP 프로토콜은 유선망의 특성을 고려하여 개발하였기 때문에 무선망에서 혼잡한 상황에서 패킷이 도달하지 못한 경우와 실제로 패킷이 손실되어 전달되지 못하는 경우를 구분하지 못한다. 최근까지 제시된 여러 이동망 TCP에 대한 논문은 무선망에서 패킷이 손실된 경우 혼잡 제어를 일어나지 못하게 하는 방법을 제시하고 있다. 본 논문에서는 TCP Persist Timer를 이용하여 혼잡제어를 회피하는 방법을 기존에 제시된 Snoop 프로토콜에 적용하여 자체적인 이동망상에서의 TCP 성능향상에 더하여 연속적인 에러에 대한 성능 향상을 제고하고 있다. 개선된 Snoop 프로토콜은 WZACK(Window Size Zero ACKnowledge Packet)을 이용하여 혼잡제어를 정지시킴으로써 비효율적인 혼잡제어를 막도록한다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.8B
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• pp.699-706
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• 2008

This paper compares throughput performance of TCP and SCTP in a variety of network environments. For experiments, we construct a Linux-based testbed and consider a set of performance metrics such as MSS(Maximum Segment Size), transmission delay, and packet loss rate. In addition, we analyze the effect of SCTP multi-streaming on throughput. From the experimental results, we can see that SCTP provides throughput gain of approximately $20%{\sim}50%$ over TCP. This performance gain comes from the distinctive features of SCTP such as chunk bundling, initial congestion window of 2 MTU and SACK(Selective ACK) based error control. In the lossy networks, we can see that SCTP multi-streaming transmissions can effectively overcome the so-called HoLB(Head-of-Line Blocking) phenomenon of TCP.